--- 1/draft-ietf-rtcweb-rtp-usage-13.txt 2014-05-16 05:14:20.470411011 -0700 +++ 2/draft-ietf-rtcweb-rtp-usage-14.txt 2014-05-16 05:14:20.562413202 -0700 @@ -1,21 +1,21 @@ RTCWEB Working Group C. Perkins Internet-Draft University of Glasgow Intended status: Standards Track M. Westerlund -Expires: October 25, 2014 Ericsson +Expires: November 17, 2014 Ericsson J. Ott Aalto University - April 23, 2014 + May 16, 2014 Web Real-Time Communication (WebRTC): Media Transport and Use of RTP - draft-ietf-rtcweb-rtp-usage-13 + draft-ietf-rtcweb-rtp-usage-14 Abstract The Web Real-Time Communication (WebRTC) framework provides support for direct interactive rich communication using audio, video, text, collaboration, games, etc. between two peers' web-browsers. This memo describes the media transport aspects of the WebRTC framework. It specifies how the Real-time Transport Protocol (RTP) is used in the WebRTC context, and gives requirements for which RTP features, profiles, and extensions need to be supported. @@ -28,21 +28,21 @@ Internet-Drafts are working documents of the Internet Engineering Task Force (IETF). Note that other groups may also distribute working documents as Internet-Drafts. The list of current Internet- Drafts is at http://datatracker.ietf.org/drafts/current/. Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference material or to cite them other than as "work in progress." - This Internet-Draft will expire on October 25, 2014. + This Internet-Draft will expire on November 17, 2014. Copyright Notice Copyright (c) 2014 IETF Trust and the persons identified as the document authors. All rights reserved. This document is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents (http://trustee.ietf.org/license-info) in effect on the date of publication of this document. Please review these documents @@ -53,69 +53,68 @@ described in the Simplified BSD License. Table of Contents 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3 2. Rationale . . . . . . . . . . . . . . . . . . . . . . . . . . 4 3. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4 4. WebRTC Use of RTP: Core Protocols . . . . . . . . . . . . . . 5 4.1. RTP and RTCP . . . . . . . . . . . . . . . . . . . . . . 5 4.2. Choice of the RTP Profile . . . . . . . . . . . . . . . . 7 - 4.3. Choice of RTP Payload Formats . . . . . . . . . . . . . . 7 + 4.3. Choice of RTP Payload Formats . . . . . . . . . . . . . . 8 4.4. Use of RTP Sessions . . . . . . . . . . . . . . . . . . . 9 - 4.5. RTP and RTCP Multiplexing . . . . . . . . . . . . . . . . 9 + 4.5. RTP and RTCP Multiplexing . . . . . . . . . . . . . . . . 10 4.6. Reduced Size RTCP . . . . . . . . . . . . . . . . . . . . 10 - 4.7. Symmetric RTP/RTCP . . . . . . . . . . . . . . . . . . . 10 + 4.7. Symmetric RTP/RTCP . . . . . . . . . . . . . . . . . . . 11 4.8. Choice of RTP Synchronisation Source (SSRC) . . . . . . . 11 - 4.9. Generation of the RTCP Canonical Name (CNAME) . . . . . . 11 - 4.10. Handling of Leap Seconds . . . . . . . . . . . . . . . . 12 - 5. WebRTC Use of RTP: Extensions . . . . . . . . . . . . . . . . 12 - 5.1. Conferencing Extensions and Topologies . . . . . . . . . 12 - 5.1.1. Full Intra Request (FIR) . . . . . . . . . . . . . . 14 - 5.1.2. Picture Loss Indication (PLI) . . . . . . . . . . . . 14 - 5.1.3. Slice Loss Indication (SLI) . . . . . . . . . . . . . 14 + 4.9. Generation of the RTCP Canonical Name (CNAME) . . . . . . 12 + 4.10. Handling of Leap Seconds . . . . . . . . . . . . . . . . 13 + 5. WebRTC Use of RTP: Extensions . . . . . . . . . . . . . . . . 13 + 5.1. Conferencing Extensions and Topologies . . . . . . . . . 13 + 5.1.1. Full Intra Request (FIR) . . . . . . . . . . . . . . 15 + 5.1.2. Picture Loss Indication (PLI) . . . . . . . . . . . . 15 + 5.1.3. Slice Loss Indication (SLI) . . . . . . . . . . . . . 15 5.1.4. Reference Picture Selection Indication (RPSI) . . . . 15 - 5.1.5. Temporal-Spatial Trade-off Request (TSTR) . . . . . . 15 + 5.1.5. Temporal-Spatial Trade-off Request (TSTR) . . . . . . 16 5.1.6. Temporary Maximum Media Stream Bit Rate Request - (TMMBR) . . . . . . . . . . . . . . . . . . . . . . . 15 + (TMMBR) . . . . . . . . . . . . . . . . . . . . . . . 16 5.2. Header Extensions . . . . . . . . . . . . . . . . . . . . 16 - 5.2.1. Rapid Synchronisation . . . . . . . . . . . . . . . . 16 - 5.2.2. Client-to-Mixer Audio Level . . . . . . . . . . . . . 16 + 5.2.1. Rapid Synchronisation . . . . . . . . . . . . . . . . 17 + 5.2.2. Client-to-Mixer Audio Level . . . . . . . . . . . . . 17 5.2.3. Mixer-to-Client Audio Level . . . . . . . . . . . . . 17 - 6. WebRTC Use of RTP: Improving Transport Robustness . . . . . . 17 - 6.1. Negative Acknowledgements and RTP Retransmission . . . . 17 - 6.2. Forward Error Correction (FEC) . . . . . . . . . . . . . 18 + 6. WebRTC Use of RTP: Improving Transport Robustness . . . . . . 18 + 6.1. Negative Acknowledgements and RTP Retransmission . . . . 18 + 6.2. Forward Error Correction (FEC) . . . . . . . . . . . . . 19 7. WebRTC Use of RTP: Rate Control and Media Adaptation . . . . 19 7.1. Boundary Conditions and Circuit Breakers . . . . . . . . 20 - 7.2. RTCP Limitations for Congestion Control . . . . . . . . . 20 - 7.3. Congestion Control Interoperability and Legacy Systems . 22 - 8. WebRTC Use of RTP: Performance Monitoring . . . . . . . . . . 23 - 9. WebRTC Use of RTP: Future Extensions . . . . . . . . . . . . 24 - 10. Signalling Considerations . . . . . . . . . . . . . . . . . . 24 - 11. WebRTC API Considerations . . . . . . . . . . . . . . . . . . 25 - 12. RTP Implementation Considerations . . . . . . . . . . . . . . 28 - 12.1. Configuration and Use of RTP Sessions . . . . . . . . . 28 - 12.1.1. Use of Multiple Media Sources Within an RTP Session 28 - 12.1.2. Use of Multiple RTP Sessions . . . . . . . . . . . . 29 - 12.1.3. Differentiated Treatment of RTP Packet Streams . . . 34 + 7.2. Congestion Control Interoperability and Legacy Systems . 21 + 8. WebRTC Use of RTP: Performance Monitoring . . . . . . . . . . 22 + 9. WebRTC Use of RTP: Future Extensions . . . . . . . . . . . . 22 + 10. Signalling Considerations . . . . . . . . . . . . . . . . . . 22 + 11. WebRTC API Considerations . . . . . . . . . . . . . . . . . . 24 + 12. RTP Implementation Considerations . . . . . . . . . . . . . . 26 + 12.1. Configuration and Use of RTP Sessions . . . . . . . . . 26 + 12.1.1. Use of Multiple Media Sources Within an RTP Session 26 + 12.1.2. Use of Multiple RTP Sessions . . . . . . . . . . . . 27 + 12.1.3. Differentiated Treatment of RTP Packet Streams . . . 32 12.2. Media Source, RTP Packet Streams, and Participant - Identification . . . . . . . . . . . . . . . . . . . . . 35 - 12.2.1. Media Source . . . . . . . . . . . . . . . . . . . . 36 - 12.2.2. SSRC Collision Detection . . . . . . . . . . . . . . 36 - 12.2.3. Media Synchronisation Context . . . . . . . . . . . 37 - 13. Security Considerations . . . . . . . . . . . . . . . . . . . 38 - 14. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 39 - 15. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 39 - 16. References . . . . . . . . . . . . . . . . . . . . . . . . . 39 - 16.1. Normative References . . . . . . . . . . . . . . . . . . 39 - 16.2. Informative References . . . . . . . . . . . . . . . . . 42 - Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 44 + Identification . . . . . . . . . . . . . . . . . . . . . 34 + 12.2.1. Media Source Identification . . . . . . . . . . . . 34 + 12.2.2. SSRC Collision Detection . . . . . . . . . . . . . . 34 + 12.2.3. Media Synchronisation Context . . . . . . . . . . . 36 + 13. Security Considerations . . . . . . . . . . . . . . . . . . . 36 + 14. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 37 + 15. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 37 + 16. References . . . . . . . . . . . . . . . . . . . . . . . . . 37 + 16.1. Normative References . . . . . . . . . . . . . . . . . . 37 + 16.2. Informative References . . . . . . . . . . . . . . . . . 40 + Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 42 1. Introduction The Real-time Transport Protocol (RTP) [RFC3550] provides a framework for delivery of audio and video teleconferencing data and other real- time media applications. Previous work has defined the RTP protocol, along with numerous profiles, payload formats, and other extensions. When combined with appropriate signalling, these form the basis for many teleconferencing systems. @@ -201,21 +200,21 @@ Bi-directional Transport-layer Flow: A bi-directional transport- layer flow is a transport-layer flow that is symmetric. That is, the transport-layer flow in the reverse direction has a 5-tuple where the source and destination address and ports are swapped compared to the forward path transport-layer flow, and the transport protocol is the same. This document uses the terminology from [I-D.ietf-avtext-rtp-grouping-taxonomy]. Other terms are used according to their definitions from the RTP Specification [RFC3550]. - We especially note the following frequently used terms: RTP Packet + Especially note the following frequently used terms: RTP Packet Stream, RTP Session, and End-point. 4. WebRTC Use of RTP: Core Protocols The following sections describe the core features of RTP and RTCP that need to be implemented, along with the mandated RTP profiles. Also described are the core extensions providing essential features that all WebRTC implementations need to implement to function effectively on today's networks. @@ -238,55 +237,71 @@ optimisations for multi-SSRC sessions defined in [I-D.ietf-avtcore-rtp-multi-stream-optimisation] is RECOMMENDED. o Random choice of SSRC on joining a session; collision detection and resolution for SSRC values (see also Section 4.8). o Support for reception of RTP data packets containing CSRC lists, as generated by RTP mixers, and RTCP packets relating to CSRCs. o Sending correct synchronisation information in the RTCP Sender - Reports, to allow receivers to implement lip-synchronisation; - support for the rapid RTP synchronisation extensions (see - Section 5.2.1) is RECOMMENDED. + Reports, to allow receivers to implement lip-synchronisation; see + Section 5.2.1 regarding support for the rapid RTP synchronisation + extensions. o Support for multiple synchronisation contexts. Participants that send multiple simultaneous RTP packet streams SHOULD do so as part of a single synchronisation context, using a single RTCP CNAME for all streams and allowing receivers to play the streams out in a synchronised manner. For compatibility with potential future versions of this specification, or for interoperability with non- WebRTC devices through a gateway, receivers MUST support multiple synchronisation contexts, indicated by the use of multiple RTCP CNAMEs in an RTP session. This specification requires the usage of a single CNAME when sending RTP Packet Streams in some circumstances, see Section 4.9. o Support for sending and receiving RTCP SR, RR, SDES, and BYE packet types, with OPTIONAL support for other RTCP packet types - unless mandated by other parts of this specification; - implementations MUST ignore unknown RTCP packet types. Note that + unless mandated by other parts of this specification. Note that additional RTCP Packet types are used by the RTP/SAVPF Profile (Section 4.2) and the other RTCP extensions (Section 5). o Support for multiple end-points in a single RTP session, and for scaling the RTCP transmission interval according to the number of participants in the session; support for randomised RTCP transmission intervals to avoid synchronisation of RTCP reports; - support for RTCP timer reconsideration. + support for RTCP timer reconsideration (Section 6.3.6 of + [RFC3550]) and reverse reconsideration (Section 6.3.4 of + [RFC3550]). o Support for configuring the RTCP bandwidth as a fraction of the media bandwidth, and for configuring the fraction of the RTCP bandwidth allocated to senders, e.g., using the SDP "b=" line - [RFC4566][RFC3556]. Support for the reduced minimum RTCP - reporting interval described in Section 6.2 of [RFC3550] is - RECOMMENDED. + [RFC4566][RFC3556]. + + o Support for the reduced minimum RTCP reporting interval described + in Section 6.2 of [RFC3550] is REQUIRED. When using the reduced + minimum RTCP reporting interval, the fixed (non-reduced) minimum + interval MUST be used when calculating the participant timeout + interval (see Sections 6.2 and 6.3.5 of [RFC3550]). The delay + before sending the initial compound RTCP packet can be set to zero + (see Section 6.2 of [RFC3550] as updated by + [I-D.ietf-avtcore-rtp-multi-stream]). + + o Ignore unknown RTCP packet types and RTP header extensions. This + to ensure robust handling of future extensions, middlebox + behaviours, etc., that can result in not signalled RTCP packet + types or RTP header extensions being received. If a compound RTCP + packet is received that contains a mixture of known and unknown + RTCP packet types, the known packets types need to be processed as + usual, with only the unknown packet types being discarded. It is known that a significant number of legacy RTP implementations, especially those targeted at VoIP-only systems, do not support all of the above features, and in some cases do not support RTCP at all. Implementers are advised to consider the requirements for graceful degradation when interoperating with legacy implementations. Other implementation considerations are discussed in Section 12. 4.2. Choice of the RTP Profile @@ -345,54 +360,60 @@ End-points can signal support for multiple RTP payload formats, or multiple configurations of a single RTP payload format, as long as each unique RTP payload format configuration uses a different RTP payload type number. As outlined in Section 4.8, the RTP payload type number is sometimes used to associate an RTP packet stream with a signalling context. This association is possible provided unique RTP payload type numbers are used in each context. For example, an RTP packet stream can be associated with an SDP "m=" line by comparing the RTP payload type numbers used by the RTP packet stream with payload types signalled in the "a=rtpmap:" lines in the media - sections of the SDP. If RTP packet streams are being associated with - signalling contexts based on the RTP payload type, then the - assignment of RTP payload type numbers MUST be unique across - signalling contexts; if the same RTP payload format configuration is - used in multiple contexts, then a different RTP payload type number - has to be assigned in each context to ensure uniqueness. If the RTP - payload type number is not being used to associate RTP packet streams - with a signalling context, then the same RTP payload type number can - be used to indicate the exact same RTP payload format configuration - in multiple contexts. A single RTP payload type number MUST NOT be - assigned to different RTP payload formats, or different - configurations of the same RTP payload format, within a single RTP - session (note that the different "m=" lines in an SDP bundle group - [I-D.ietf-mmusic-sdp-bundle-negotiation] form a single RTP session). + sections of the SDP. This leads to the following considerations: + + If RTP packet streams are being associated with signalling + contexts based on the RTP payload type, then the assignment of RTP + payload type numbers MUST be unique across signalling contexts. + + If the same RTP payload format configuration is used in multiple + contexts, then a different RTP payload type number has to be + assigned in each context to ensure uniqueness. + + If the RTP payload type number is not being used to associate RTP + packet streams with a signalling context, then the same RTP + payload type number can be used to indicate the exact same RTP + payload format configuration in multiple contexts. + + A single RTP payload type number MUST NOT be assigned to different + RTP payload formats, or different configurations of the same RTP + payload format, within a single RTP session (note that the "m=" lines + in an SDP bundle group [I-D.ietf-mmusic-sdp-bundle-negotiation] form + a single RTP session). An end-point that has signalled support for multiple RTP payload - formats SHOULD be able to accept data in any of those payload formats + formats MUST be able to accept data in any of those payload formats at any time, unless it has previously signalled limitations on its decoding capability. This requirement is constrained if several types of media (e.g., audio and video) are sent in the same RTP session. In such a case, a source (SSRC) is restricted to switching only between the RTP payload formats signalled for the type of media that is being sent by that source; see Section 4.4. To support rapid rate adaptation by changing codec, RTP does not require advance signalling for changes between RTP payload formats used by a single SSRC that were signalled during session set-up. - An RTP sender that changes between two RTP payload types that use - different RTP clock rates MUST follow the recommendations in - Section 4.1 of [RFC7160]. RTP receivers MUST follow the - recommendations in Section 4.3 of [RFC7160] in order to support - sources that switch between clock rates in an RTP session (these - recommendations for receivers are backwards compatible with the case - where senders use only a single clock rate). + If performing changes between two RTP payload types that use + different RTP clock rates, an RTP sender MUST follow the + recommendations in Section 4.1 of [RFC7160]. RTP receivers MUST + follow the recommendations in Section 4.3 of [RFC7160] in order to + support sources that switch between clock rates in an RTP session + (these recommendations for receivers are backwards compatible with + the case where senders use only a single clock rate). 4.4. Use of RTP Sessions An association amongst a set of end-points communicating using RTP is known as an RTP session [RFC3550]. An end-point can be involved in several RTP sessions at the same time. In a multimedia session, each type of media has typically been carried in a separate RTP session (e.g., using one RTP session for the audio, and a separate RTP session using a different transport-layer flow for the video). WebRTC implementations of RTP are REQUIRED to implement support for @@ -410,96 +431,99 @@ REQUIRED to support transport of all RTP packet streams, independent of media type, in a single RTP session using a single transport layer flow, according to [I-D.ietf-avtcore-multi-media-rtp-session]. If multiple types of media are to be used in a single RTP session, all participants in that RTP session MUST agree to this usage. In an SDP context, [I-D.ietf-mmusic-sdp-bundle-negotiation] can be used to signal such a bundle of RTP packet streams forming a single RTP session. Further discussion about the suitability of different RTP session - structures and multiplexing methods to different scenarios are - suitable can be found in [I-D.ietf-avtcore-multiplex-guidelines]. + structures and multiplexing methods to different scenarios can be + found in [I-D.ietf-avtcore-multiplex-guidelines]. 4.5. RTP and RTCP Multiplexing Historically, RTP and RTCP have been run on separate transport layer flows (e.g., two UDP ports for each RTP session, one port for RTP and one port for RTCP). With the increased use of Network Address/Port Translation (NAT/NAPT) this has become problematic, since maintaining multiple NAT bindings can be costly. It also complicates firewall administration, since multiple ports need to be opened to allow RTP - traffic. To reduce these costs and session set-up times, support for - multiplexing RTP data packets and RTCP control packets on a single - transport-layer flow for each RTP session is REQUIRED, provided it is - negotiated in the signalling channel before use as specified in - [RFC5761]. For backwards compatibility, implementations are also - REQUIRED to support RTP and RTCP sent on separate transport-layer - flows. + traffic. To reduce these costs and session set-up times, + implementations are REQUIRED to support multiplexing RTP data packets + and RTCP control packets on a single transport-layer flow [RFC5761]. + Such RTP and RTCP multiplexing MUST be negotiated in the signalling + channel before it is used. If SDP is used for signalling, this + negotiation MUST use the attributes defined in [RFC5761]. For + backwards compatibility, implementations are also REQUIRED to support + RTP and RTCP sent on separate transport-layer flows. Note that the use of RTP and RTCP multiplexed onto a single transport-layer flow ensures that there is occasional traffic sent on that port, even if there is no active media traffic. This can be - useful to keep NAT bindings alive, and is the recommend method for - application level keep-alives of RTP sessions [RFC6263]. + useful to keep NAT bindings alive [RFC6263]. 4.6. Reduced Size RTCP RTCP packets are usually sent as compound RTCP packets, and [RFC3550] requires that those compound packets start with an Sender Report (SR) or Receiver Report (RR) packet. When using frequent RTCP feedback messages under the RTP/AVPF Profile [RFC4585] these statistics are not needed in every packet, and unnecessarily increase the mean RTCP packet size. This can limit the frequency at which RTCP packets can be sent within the RTCP bandwidth share. To avoid this problem, [RFC5506] specifies how to reduce the mean RTCP message size and allow for more frequent feedback. Frequent feedback, in turn, is essential to make real-time applications quickly aware of changing network conditions, and to allow them to - adapt their transmission and encoding behaviour. Support for non- - compound RTCP feedback packets [RFC5506] is REQUIRED, but MUST be - negotiated using the signalling channel before use. For backwards - compatibility, implementations are also REQUIRED to support the use - of compound RTCP feedback packets if the remote end-point does not - agree to the use of non-compound RTCP in the signalling exchange. + adapt their transmission and encoding behaviour. Implementations + MUST support sending and receiving non-compound RTCP feedback packets + [RFC5506]. Use of non-compound RTCP packets MUST be negotiated using + the signalling channel. If SDP is used for signalling, this + negotiation MUST use the attributes defined in [RFC5506]. For + backwards compatibility, implementations are also REQUIRED to support + the use of compound RTCP feedback packets if the remote end-point + does not agree to the use of non-compound RTCP in the signalling + exchange. 4.7. Symmetric RTP/RTCP To ease traversal of NAT and firewall devices, implementations are REQUIRED to implement and use Symmetric RTP [RFC4961]. The reason for using symmetric RTP is primarily to avoid issues with NATs and Firewalls by ensuring that the send and receive RTP packet streams, as well as RTCP, are actually bi-directional transport-layer flows. This will keep alive the NAT and firewall pinholes, and help indicate consent that the receive direction is a transport-layer flow the intended recipient actually wants. In addition, it saves resources, specifically ports at the end-points, but also in the network as NAT mappings or firewall state is not unnecessary bloated. The amount of per flow QoS state kept in the network is also reduced. 4.8. Choice of RTP Synchronisation Source (SSRC) Implementations are REQUIRED to support signalled RTP synchronisation - source (SSRC) identifiers, using the "a=ssrc:" SDP attribute defined - in Section 4.1 and Section 5 of [RFC5576]. Implementations MUST also - support the "previous-ssrc" source attribute defined in Section 6.2 - of [RFC5576]. Other per-SSRC attributes defined in [RFC5576] MAY be - supported. + source (SSRC) identifiers. If SDP is used, this MUST be done using + the "a=ssrc:" SDP attribute defined in Section 4.1 and Section 5 of + [RFC5576] and the "previous-ssrc" source attribute defined in + Section 6.2 of [RFC5576]; other per-SSRC attributes defined in + [RFC5576] MAY be supported. - Use of the "a=ssrc:" attribute to signal SSRC identifiers in an RTP - session is OPTIONAL. Implementations MUST be prepared to accept RTP - and RTCP packets using SSRCs that have not been explicitly signalled - ahead of time. Implementations MUST support random SSRC assignment, - and MUST support SSRC collision detection and resolution, according - to [RFC3550]. When using signalled SSRC values, collision detection - MUST be performed as described in Section 5 of [RFC5576]. + While support for signalled SSRC identifiers is mandated, their use + in an RTP session is OPTIONAL. Implementations MUST be prepared to + accept RTP and RTCP packets using SSRCs that have not been explicitly + signalled ahead of time. Implementations MUST support random SSRC + assignment, and MUST support SSRC collision detection and resolution, + according to [RFC3550]. When using signalled SSRC values, collision + detection MUST be performed as described in Section 5 of [RFC5576]. It is often desirable to associate an RTP packet stream with a non- RTP context. For users of the WebRTC API a mapping between SSRCs and MediaStreamTracks are provided per Section 11. For gateways or other usages it is possible to associate an RTP packet stream with an "m=" line in a session description formatted using SDP. If SSRCs are signalled this is straightforward (in SDP the "a=ssrc:" line will be at the media level, allowing a direct association with an "m=" line). If SSRCs are not signalled, the RTP payload type numbers used in an RTP packet stream are often sufficient to associate that packet @@ -548,21 +572,21 @@ RTCPeerConnection within their common same-origin context. An WebRTC end-point MUST support reception of any CNAME that matches the syntax limitations specified by the RTP specification [RFC3550] and cannot assume that any CNAME will be chosen according to the form suggested above. 4.10. Handling of Leap Seconds The guidelines regarding handling of leap seconds to limit their - impact on RTP media playout and synchronization given in [RFC7164] + impact on RTP media play-out and synchronization given in [RFC7164] SHOULD be followed. 5. WebRTC Use of RTP: Extensions There are a number of RTP extensions that are either needed to obtain full functionality, or extremely useful to improve on the baseline performance, in the WebRTC application context. One set of these extensions is related to conferencing, while others are more generic in nature. The following subsections describe the various RTP extensions mandated or suggested for use within the WebRTC context. @@ -580,23 +604,23 @@ While the use of IP multicast groups is popular in IPTV systems, the topologies based on RTP middleboxes are dominant in interactive video conferencing environments. Topologies based on a mesh of unicast transport-layer flows to create a common RTP session have not seen widespread deployment to date. Accordingly, WebRTC implementations are not expected to support topologies based on IP multicast groups or to support mesh-based topologies, such as a point-to-multipoint mesh configured as a single RTP session (Topo-Mesh in the terminology of [I-D.ietf-avtcore-rtp-topologies-update]). However, a point-to- multipoint mesh constructed using several RTP sessions, implemented - in the WebRTC context using independent RTCPeerConnections, can be - expected to be utilised by WebRTC applications and needs to be - supported. + in the WebRTC context using independent RTCPeerConnections + [W3C.WD-webrtc-20130910], can be expected to be utilised by WebRTC + applications and needs to be supported. WebRTC implementations of RTP endpoints implemented according to this memo are expected to support all the topologies described in [I-D.ietf-avtcore-rtp-topologies-update] where the RTP endpoints send and receive unicast RTP packet streams to and from some peer device, provided that peer can participate in performing congestion control on the RTP packet streams. The peer device could be another RTP endpoint, or it could be an RTP middlebox that redistributes the RTP packet streams to other RTP endpoints. This limitation means that some of the RTP middlebox-based topologies are not suitable for use @@ -641,21 +665,21 @@ profile (RTP/SAVPF) [RFC5124]. 5.1.1. Full Intra Request (FIR) The Full Intra Request message is defined in Sections 3.5.1 and 4.3.1 of the Codec Control Messages [RFC5104]. It is used to make the mixer request a new Intra picture from a participant in the session. This is used when switching between sources to ensure that the receivers can decode the video or other predictive media encoding with long prediction chains. WebRTC senders MUST understand and - react to FIR feedback messages they receiver, since this greatly + react to FIR feedback messages they receive, since this greatly improves the user experience when using centralised mixer-based conferencing. Support for sending FIR messages is OPTIONAL. 5.1.2. Picture Loss Indication (PLI) The Picture Loss Indication message is defined in Section 6.3.1 of the RTP/AVPF profile [RFC4585]. It is used by a receiver to tell the sending encoder that it lost the decoder context and would like to have it repaired somehow. This is semantically different from the Full Intra Request above as there could be multiple ways to fulfil @@ -748,91 +772,91 @@ general RTP header extension mechanism [RFC5285], which requires signalling, but are otherwise backwards compatible. 5.2.2. Client-to-Mixer Audio Level The Client to Mixer Audio Level extension [RFC6464] is an RTP header extension used by an endpoint to inform a mixer about the level of audio activity in the packet to which the header is attached. This enables an RTP middlebox to make mixing or selection decisions without decoding or detailed inspection of the payload, reducing the - complexity in some types of mixer. It can also save decoding + complexity in some types of mixers. It can also save decoding resources in receivers, which can choose to decode only the most relevant RTP packet streams based on audio activity levels. The Client-to-Mixer Audio Level [RFC6464] header extension is RECOMMENDED to be implemented. If this header extension is implemented, it is REQUIRED that implementations are capable of encrypting the header extension according to [RFC6904] since the information contained in these header extensions can be considered - sensitive. It is further RECOMMENDED that this encryption is used, - unless the encryption has been explicitly disabled through API or + sensitive. The use of this encryption is RECOMMENDED, however usage + of the encryption can be explicitly disabled through API or signalling. 5.2.3. Mixer-to-Client Audio Level The Mixer to Client Audio Level header extension [RFC6465] provides an endpoint with the audio level of the different sources mixed into - a common mix by a RTP mixer. This enables a user interface to - indicate the relative activity level of each session participant, + a common source stream by a RTP mixer. This enables a user interface + to indicate the relative activity level of each session participant, rather than just being included or not based on the CSRC field. This - is a pure optimisations of non critical functions, and is hence + is a pure optimisation of non critical functions, and is hence OPTIONAL to implement. If this header extension is implemented, it is REQUIRED that implementations are capable of encrypting the header extension according to [RFC6904] since the information contained in these header extensions can be considered sensitive. It is further RECOMMENDED that this encryption is used, unless the encryption has been explicitly disabled through API or signalling. 6. WebRTC Use of RTP: Improving Transport Robustness There are tools that can make RTP packet streams robust against packet loss and reduce the impact of loss on media quality. However, - they all add overhead compared to a non-robust stream. The overhead - needs to be considered, and the aggregate bit-rate MUST be rate - controlled to avoid causing network congestion (see Section 7). As a - result, improving robustness might require a lower base encoding - quality, but has the potential to deliver that quality with fewer - errors. The mechanisms described in the following sub-sections can - be used to improve tolerance to packet loss. + they generally some add overhead compared to a non-robust stream. + The overhead needs to be considered, and the aggregate bit-rate MUST + be rate controlled to avoid causing network congestion (see + Section 7). As a result, improving robustness might require a lower + base encoding quality, but has the potential to deliver that quality + with fewer errors. The mechanisms described in the following sub- + sections can be used to improve tolerance to packet loss. 6.1. Negative Acknowledgements and RTP Retransmission As a consequence of supporting the RTP/SAVPF profile, implementations can send negative acknowledgements (NACKs) for RTP data packets [RFC4585]. This feedback can be used to inform a sender of the loss of particular RTP packets, subject to the capacity limitations of the RTCP feedback channel. A sender can use this information to optimise the user experience by adapting the media encoding to compensate for known lost packets. - RTP packet stream Senders are REQUIRED to understand the Generic NACK + RTP packet stream senders are REQUIRED to understand the Generic NACK message defined in Section 6.2.1 of [RFC4585], but MAY choose to ignore some or all of this feedback (following Section 4.2 of [RFC4585]). Receivers MAY send NACKs for missing RTP packets. Guidelines on when to send NACKs are provided in [RFC4585]. It is not expected that a receiver will send a NACK for every lost RTP packet, rather it needs to consider the cost of sending NACK feedback, and the importance of the lost packet, to make an informed decision on whether it is worth telling the sender about a packet loss event. The RTP Retransmission Payload Format [RFC4588] offers the ability to retransmit lost packets based on NACK feedback. Retransmission needs to be used with care in interactive real-time applications to ensure that the retransmitted packet arrives in time to be useful, but can be effective in environments with relatively low network RTT (an RTP sender can estimate the RTT to the receivers using the information in RTCP SR and RR packets, as described at the end of Section 6.4.1 of [RFC3550]). The use of retransmissions can also increase the forward RTP bandwidth, and can potentially caused increased packet loss if - the original packet loss was caused by network congestion. We note, + the original packet loss was caused by network congestion. Note, however, that retransmission of an important lost packet to repair decoder state can have lower cost than sending a full intra frame. It is not appropriate to blindly retransmit RTP packets in response to a NACK. The importance of lost packets and the likelihood of them arriving in time to be useful needs to be considered before RTP retransmission is used. Receivers are REQUIRED to implement support for RTP retransmission packets [RFC4588]. Senders MAY send RTP retransmission packets in response to NACKs if the RTP retransmission payload format has been @@ -892,158 +916,53 @@ bandwidth, or it might be competition with other traffic on the link (this can be non-WebRTC traffic, traffic due to other WebRTC flows, or even competition with other WebRTC flows in the same session). An effective media congestion control algorithm is therefore an essential part of the WebRTC framework. However, at the time of this writing, there is no standard congestion control algorithm that can be used for interactive media applications such as WebRTC's flows. Some requirements for congestion control algorithms for RTCPeerConnections are discussed in [I-D.ietf-rmcat-cc-requirements]. - It is expected that a future version of this memo will mandate the - use of a congestion control algorithm that satisfies these - requirements. + A future version of this memo will mandate the use of a congestion + control algorithm that satisfies these requirements. 7.1. Boundary Conditions and Circuit Breakers - In the absence of a concrete congestion control algorithm, all WebRTC - implementations MUST implement the RTP circuit breaker algorithm that - is described in [I-D.ietf-avtcore-rtp-circuit-breakers]. The RTP - circuit breaker is designed to enable applications to recognise and - react to situations of extreme network congestion. However, since - the RTP circuit breaker might not be triggered until congestion - becomes extreme, it cannot be considered a substitute for congestion - control, and applications MUST also implement congestion control to - allow them to adapt to changes in network capacity. Any future RTP - congestion control algorithms are expected to operate within the - envelope allowed by the circuit breaker. + WebRTC implementations MUST implement the RTP circuit breaker + algorithm that is described in + [I-D.ietf-avtcore-rtp-circuit-breakers]. The RTP circuit breaker is + designed to enable applications to recognise and react to situations + of extreme network congestion. However, since the RTP circuit + breaker might not be triggered until congestion becomes extreme, it + cannot be considered a substitute for congestion control, and + applications MUST also implement congestion control to allow them to + adapt to changes in network capacity. Any future RTP congestion + control algorithms are expected to operate within the envelope + allowed by the circuit breaker. The session establishment signalling will also necessarily establish boundaries to which the media bit-rate will conform. The choice of media codecs provides upper- and lower-bounds on the supported bit- rates that the application can utilise to provide useful quality, and - the packetization choices that exist. In addition, the signalling - channel can establish maximum media bit-rate boundaries using the SDP - "b=AS:" or "b=CT:" lines, and the RTP/AVPF Temporary Maximum Media - Stream Bit Rate (TMMBR) Requests (see Section 5.1.6 of this memo). - The combination of media codec choice and signalled bandwidth limits - SHOULD be used to limit traffic based on known bandwidth limitations, - for example the capacity of the edge links, to the extent possible. - -7.2. RTCP Limitations for Congestion Control - - Experience with the congestion control algorithms of TCP [RFC5681], - TFRC [RFC5348], and DCCP [RFC4341], [RFC4342], [RFC4828], has shown - that feedback on packet arrivals needs to be sent frequently (roughly - once per round trip time is common). We note that the real-time - media traffic might not be able to adapt to changing path conditions - as rapidly as elastic applications using TCP, but frequent feedback, - perhaps on the order of once per video frame, is still needed to - allow the congestion control algorithm to track the path dynamics. - - As an example of the type of RTCP congestion control feedback that is - possible, consider one of the simplest scenarios for WebRTC: a point - to point video call between two end systems. There will be four RTP - flows in this scenario, two audio and two video, with all four flows - being active for essentially all the time (the audio flows will - likely use voice activity detection and comfort noise to reduce the - packet rate during silent periods, but doesn't cause transmissions to - stop). Assume all four flows are sent in a single RTP session, each - using a separate SSRC. Further, assume each SSRC sends RTCP reports - for all other SSRCs in the session (i.e., the optimisations in - [I-D.ietf-avtcore-rtp-multi-stream-optimisation] are not used, giving - the worst case for the RTCP overhead). When all members are senders - like this, the RTCP timing rules in Sections 6.2 and 6.3 of [RFC3550] - and [RFC4585] reduce to: - - rtcp_interval = avg_rtcp_size * n / rtcp_bw - - where avg_rtcp_size is measured in octets, and the rtcp_bw is the - bandwidth available for RTCP. The average RTCP size will depend on - the amount of feedback that is sent in each RTCP packet, on the - number of members in the session, and on the size of source - description (RTCP SDES) information sent. As a baseline, each RTCP - packet will be a compound RTCP packet that contains an RTCP SR and an - RTCP SDES packet. In the scenario above, each RTCP SR packet will - contain three report blocks, once for each of the other RTP SSRCs - sending data, for a total of 100 octets (this is 8 octets header, 20 - octets sender info, and 3 * 24 octets report blocks). The RTCP SDES - packet will comprise a header (4 octets), an originating SSRC (4 - octets), a CNAME chunk, and padding. If the CNAME follows [RFC7022] - and it will be 19 octets in size, and require 1 octet of padding. - The resulting compound RTCP packet will be 128 octets in size. If - sent in UDP/IPv4 with no IP options and using Secure RTP, which adds - 20 (IPv4) + 8 (UDP) + 14 (SRTP with 80 bit Authentication tag), the - avg_rtcp_size will therefore be 170 octets, including the header - overhead. The value n is this scenario is 4, and the rtcp_bw is - assumed to be 5% of the session bandwidth. - - If it is desired to send RTCP feedback packets on average 30 times - per second, to correspond to one RTCP report every frame for 30fps - video, we can invert the above rtcp_interval calculation to get an - rtcp_bw that gives an interval of 1/30th of a second or lower. This - corresponds to an rtcp_bw of 20400 octets per second (since 1/30 = - 170 * 4 / 20400). This is 163200 bits per second, which if 5% of the - session bandwidth, gives a session bandwidth of approximately 3.3Mbps - (i.e., 3.3Mbps media rate, plus an additional 5% for RTCP, to give a - total data rate of approximately 3.4Mbps). That is, RTCP can report - on every frame of video provided the session bandwidth is 3.3Mbps or - larger, when every SSRC sends a report for every video frame. Please - note that the actual RTCP transmission intervals will be within the - interval [0.0135, 0.0406]s, but maintaining an average RTCP - transmission interval of 0.033s. - - Note: To achieve the RTCP transmission intervals above the RTP/ - SAVPF profile with T_rr_interval=0 is used, since even when using - the reduced minimal transmission interval, the RTP/SAVP profile - would only allow sending RTCP at most every 0.11s (every third - frame of video). Using RTP/SAVPF with T_rr_interval=0 however is - capable of fully utilizing the configured 5% RTCP bandwidth - fraction. - - If additional feedback beyond the standard report block is needed, - the session bandwidth needed will increase. For example, with an - additional 20 octets data being reported in each RTCP packet, the - session bandwidth needed increases to 3.5Mbps for every SSRC to be - able to report on every frame. However, the above baseline might not - be the most appropriate usage of the RTCP bandwidth. Depending on - needs, a less frequent usage of regular RTCP compound packets, - controlled by T_rr_interval combined with using the reduced size RTCP - packets, can achieve more frequent and useful reporting. Also the - reporting requirements defined in - [I-D.ietf-avtcore-rtp-multi-stream-optimisation] will reduced the - amount of bandwidth consumed for reporting when each endpoint has - multiple SSRCs. - - Calculations such as these show that RTCP cannot be used to send per- - packet congestion feedback. RTCP can, however, be used to send - congestion feedback on each frame of video sent in an interactive - video conferencing scenario, provided the RTCP parameters are - correctly configured and the overall session bandwidth exceeds a - couple of megabits per second (the exact rate depending on the number - of session participants, the RTCP bandwidth fraction, and whether - audio and video are sent in one or two RTP sessions). Using similar - calculations, it can be shown that RTCP can likely also be used to - send feedback on a per-RTT basis, provided the RTT is not too low. - - Interactive communication might not be able to afford to wait for - packet losses to occur to indicate congestion, because an increase in - play out delay due to queuing (most prominent in wireless networks) - can easily lead to packets being dropped due to late arrival at the - receiver. Therefore, more sophisticated cues might need to be - reported -- to be defined in a suitable congestion control framework - as noted above -- which, in turn, increase the report size again. - For example, different RTCP XR report blocks (jointly) provide the - necessary details to implement a variety of congestion control - algorithms, but the (compound) report size grows quickly. + the packetisation choices that exist. In addition, the signalling + channel can establish maximum media bit-rate boundaries using, for + example, the SDP "b=AS:" or "b=CT:" lines and the RTP/AVPF Temporary + Maximum Media Stream Bit Rate (TMMBR) Requests (see Section 5.1.6 of + this memo). Signalled bandwidth limitations, such as SDP "b=AS:" or + "b=CT:" lines received from the peer, MUST be followed when sending + RTP packet streams. A WebRTC endpoint receiving media SHOULD signal + its bandwidth limitations, these limitations have to be based on + known bandwidth limitations, for example the capacity of the edge + links. -7.3. Congestion Control Interoperability and Legacy Systems +7.2. Congestion Control Interoperability and Legacy Systems There are legacy RTP implementations that do not implement RTCP, and hence do not provide any congestion feedback. Congestion control cannot be performed with these end-points. WebRTC implementations that need to interwork with such end-points MUST limit their transmission to a low rate, equivalent to a VoIP call using a low bandwidth codec, that is unlikely to cause any significant congestion. When interworking with legacy implementations that support RTCP using @@ -1066,22 +985,22 @@ With proprietary congestion control algorithms issues can arise when different algorithms and implementations interact in a communication session. If the different implementations have made different choices in regards to the type of adaptation, for example one sender based, and one receiver based, then one could end up in situation where one direction is dual controlled, when the other direction is not controlled. This memo cannot mandate behaviour for proprietary congestion control algorithms, but implementations that use such algorithms ought to be aware of this issue, and try to ensure that - both effective congestion control is negotiated for media flowing in - both directions. If the IETF were to standardise both sender- and + effective congestion control is negotiated for media flowing in both + directions. If the IETF were to standardise both sender- and receiver-based congestion control algorithms for WebRTC traffic in the future, the issues of interoperability, control, and ensuring that both directions of media flow are congestion controlled would also need to be considered. 8. WebRTC Use of RTP: Performance Monitoring As described in Section 4.1, implementations are REQUIRED to generate RTCP Sender Report (SR) and Reception Report (RR) packets relating to the RTP packet streams they send and receive. These RTCP reports can @@ -1090,25 +1009,20 @@ A large number of additional performance metrics are supported by the RTCP Extended Reports (XR) framework [RFC3611][RFC6792]. At the time of this writing, it is not clear what extended metrics are suitable for use in the WebRTC context, so there is no requirement that implementations generate RTCP XR packets. However, implementations that can use detailed performance monitoring data MAY generate RTCP XR packets as appropriate; the use of such packets SHOULD be signalled in advance. - All WebRTC implementations MUST be prepared to receive RTP XR report - packets, whether or not they were signalled. There is no requirement - that the data contained in such reports be used, or exposed to the - Javascript application, however. - 9. WebRTC Use of RTP: Future Extensions It is possible that the core set of RTP protocols and RTP extensions specified in this memo will prove insufficient for the future needs of WebRTC applications. In this case, future updates to this memo MUST be made following the Guidelines for Writers of RTP Payload Format Specifications [RFC2736], How to Write an RTP Payload Format [I-D.ietf-payload-rtp-howto] and Guidelines for Extending the RTP Control Protocol [RFC5968], and SHOULD take into account any future guidelines for extending RTP and related protocols that have been @@ -1130,65 +1044,64 @@ parameters: RTP Profile: The name of the RTP profile to be used in session. The RTP/AVP [RFC3551] and RTP/AVPF [RFC4585] profiles can interoperate on basic level, as can their secure variants RTP/SAVP [RFC3711] and RTP/SAVPF [RFC5124]. The secure variants of the profiles do not directly interoperate with the non-secure variants, due to the presence of additional header fields for authentication in SRTP packets and cryptographic transformation of the payload. WebRTC requires the use of the RTP/SAVPF profile, and this MUST be - signalled if SDP is used. Interworking functions might transform - this into the RTP/SAVP profile for a legacy use case, by - indicating to the WebRTC end-point that the RTP/SAVPF is used, and - limiting the usage of the "a=rtcp-fb:" attribute to indicate a - trr-int value of 4 seconds. + signalled. Interworking functions might transform this into the + RTP/SAVP profile for a legacy use case, by indicating to the + WebRTC end-point that the RTP/SAVPF is used and configuring a trr- + int value of 4 seconds. Transport Information: Source and destination IP address(s) and ports for RTP and RTCP MUST be signalled for each RTP session. In - WebRTC these transport addresses will be provided by ICE that - signals candidates and arrives at nominated candidate address + WebRTC these transport addresses will be provided by ICE [RFC5245] + that signals candidates and arrives at nominated candidate address pairs. If RTP and RTCP multiplexing [RFC5761] is to be used, such that a single port, i.e. transport-layer flow, is used for RTP and RTCP flows, this MUST be signalled (see Section 4.5). RTP Payload Types, media formats, and format parameters: The mapping between media type names (and hence the RTP payload formats to be used), and the RTP payload type numbers MUST be signalled. Each media type MAY also have a number of media type parameters that MUST also be signalled to configure the codec and RTP payload format (the "a=fmtp:" line from SDP). Section 4.3 of this memo discusses requirements for uniqueness of payload types. - RTP Extensions: The RTP extensions to be used SHOULD be agreed upon, - including any parameters for each respective extension. At the - very least, this will help avoiding using bandwidth for features - that the other end-point will ignore. But for certain mechanisms - there is requirement for this to happen as interoperability - failure otherwise happens. + RTP Extensions: The use of any additional RTP header extensions and + RTCP packet types, including any necessary parameters, SHOULD be + signalled. For robustness, and for compatibility with non-WebRTC + systems that might be connected to a WebRTC session via a gateway, + implementations are required to ignore unknown RTCP packets and + RTP header extensions (See Section 4.1). RTCP Bandwidth: Support for exchanging RTCP Bandwidth values to the end-points will be necessary. This SHALL be done as described in "Session Description Protocol (SDP) Bandwidth Modifiers for RTP - Control Protocol (RTCP) Bandwidth" [RFC3556], or something - semantically equivalent. This also ensures that the end-points - have a common view of the RTCP bandwidth, this is important as too - different view of the bandwidths can lead to failure to - interoperate. + Control Protocol (RTCP) Bandwidth" [RFC3556] if using SDP, or + something semantically equivalent. This also ensures that the + end-points have a common view of the RTCP bandwidth. A common + RTCP bandwidth is important as a too different view of the + bandwidths can lead to failure to interoperate. These parameters are often expressed in SDP messages conveyed within an offer/answer exchange. RTP does not depend on SDP or on the offer /answer model, but does require all the necessary parameters to be - agreed upon, and provided to the RTP implementation. We note that in + agreed upon, and provided to the RTP implementation. Note that in the WebRTC context it will depend on the signalling model and API how these parameters need to be configured but they will be need to - either set in the API or explicitly signalled between the peers. + either be set in the API or explicitly signalled between the peers. 11. WebRTC API Considerations The WebRTC API [W3C.WD-webrtc-20130910] and the Media Capture and Streams API [W3C.WD-mediacapture-streams-20130903] defines and uses the concept of a MediaStream that consists of zero or more MediaStreamTracks. A MediaStreamTrack is an individual stream of media from any type of media source like a microphone or a camera, but also conceptual sources, like a audio mix or a video composition, are possible. The MediaStreamTracks within a MediaStream need to be @@ -1206,96 +1119,97 @@ the source packet stream. It is important to note that the same media source can be feeding multiple MediaStreamTracks. As different sets of constraints or other parameters can be applied to the MediaStreamTrack, each MediaStreamTrack instance added to a RTCPeerConnection SHALL result in an independent source packet stream, with its own set of associated packet streams, and thus different SSRC(s). It will depend on applied constraints and parameters if the source stream and the encoding configuration will be identical between different - MediaStreamTracks sharing the same media source. Thus it is possible - for multiple source packet streams to share encoded streams (but not - packet streams), but this is an implementation choice to try to - utilise such optimisations. Note that such optimizations would need - to take into account that the constraints for one of the - MediaStreamTracks can at any moment change, meaning that the encoding - configurations might no longer be identical. + MediaStreamTracks sharing the same media source. If the encoding + parameters and constraints are the same, an implementation could + choose to use only one encoded stream to create the different RTP + packet streams. Note that such optimisations would need to take into + account that the constraints for one of the MediaStreamTracks can at + any moment change, meaning that the encoding configurations might no + longer be identical and two different encoder instances would then be + needed. The same MediaStreamTrack can also be included in multiple MediaStreams, thus multiple sets of MediaStreams can implicitly need to use the same synchronisation base. To ensure that this works in - all cases, and don't forces a end-point to change synchronisation - base and CNAME in the middle of a ongoing delivery of any packet - streams, which would cause media disruption; all MediaStreamTracks - and their associated SSRCs originating from the same end-point needs - to be sent using the same CNAME within one RTCPeerConnection. This - is motivating the strong recommendation in Section 4.9 to only use a - single CNAME. + all cases, and does not force an end-point to to disrupt the media by + changing synchronisation base and CNAME during delivery of any + ongoing packet streams, all MediaStreamTracks and their associated + SSRCs originating from the same end-point need to be sent using the + same CNAME within one RTCPeerConnection. This is motivating the + strong recommendation in Section 4.9 to only use a single CNAME. The requirement on using the same CNAME for all SSRCs that - originates from the same end-point, does not require middleboxes + originate from the same end-point, does not require a middlebox that forwards traffic from multiple end-points to only use a single CNAME. Different CNAMEs normally need to be used for different RTCPeerConnection instances, as specified in Section 4.9. Having two communication sessions with the same CNAME could enable tracking of a user or device across different services (see Section 4.4.1 of [I-D.ietf-rtcweb-security] for details). A web application can - request that the CNAMEs used in different RTCPeerConnection within a - same-orign context to be the same, this allow for synchronization of + request that the CNAMEs used in different RTCPeerConnections (within + a same-orign context) be the same, this allows for synchronization of the endpoint's RTP packet streams across the different RTCPeerConnections. Note: this doesn't result in a tracking issue, since the creation of matching CNAMEs depends on existing tracking. - The above will currently force a WebRTC end-point that receives an + The above will currently force a WebRTC end-point that receives a MediaStreamTrack on one RTCPeerConnection and adds it as an outgoing on any RTCPeerConnection to perform resynchronisation of the stream. - This, as the sending party needs to change the CNAME, which implies - that it has to use a locally available system clock as timebase for - the synchronisation. Thus, the relative relation between the - timebase of the incoming stream and the system sending out needs to - defined. This relation also needs monitoring for clock drift and - likely adjustments of the synchronisation. The sending entity is - also responsible for congestion control for its the sent streams. In - cases of packet loss the loss of incoming data also needs to be - handled. This leads to the observation that the method that is least - likely to cause issues or interruptions in the outgoing source packet - stream is a model of full decoding, including repair etc followed by - encoding of the media again into the outgoing packet stream. - Optimisations of this method is clearly possible and implementation - specific. + This, as the sending party needs to change the CNAME to the one it + uses, which implies that the sender has to use a local system clock + as timebase for the synchronisation. Thus, the relative relation + between the timebase of the incoming stream and the system sending + out needs to defined. This relation also needs monitoring for clock + drift and likely adjustments of the synchronisation. The sending + entity is also responsible for congestion control for its sent + streams. In cases of packet loss the loss of incoming data also + needs to be handled. This leads to the observation that the method + that is least likely to cause issues or interruptions in the outgoing + source packet stream is a model of full decoding, including repair + etc., followed by encoding of the media again into the outgoing + packet stream. Optimisations of this method is clearly possible and + implementation specific. A WebRTC end-point MUST support receiving multiple MediaStreamTracks, where each of different MediaStreamTracks (and their sets of associated packet streams) uses different CNAMEs. However, MediaStreamTracks that are received with different CNAMEs have no defined synchronisation. Note: The motivation for supporting reception of multiple CNAMEs - are to allow for forward compatibility with any future changes - that enables more efficient stream handling when end-points relay/ + is to allow for forward compatibility with any future changes that + enables more efficient stream handling when end-points relay/ forward streams. It also ensures that end-points can interoperate with certain types of multi-stream middleboxes or end-points that are not WebRTC. The binding between the WebRTC MediaStreams, MediaStreamTracks and the SSRC is done as specified in "Cross Session Stream Identification in the Session Description Protocol" [I-D.ietf-mmusic-msid]. This document [I-D.ietf-mmusic-msid] also defines, in section 4.1, how to map unknown source packet stream SSRCs to MediaStreamTracks and - MediaStreams. Commonly the RTP Payload Type of any incoming packets - will reveal if the packet stream is a source stream or a redundancy - or dependent packet stream. The association to the correct source + MediaStreams. This later is relevant to handle some cases of legacy + interop. Commonly the RTP Payload Type of any incoming packets will + reveal if the packet stream is a source stream or a redundancy or + dependent packet stream. The association to the correct source packet stream depends on the payload format in use for the packet stream. Finally this specification puts a requirement on the WebRTC API to realize a method for determining the CSRC list (Section 4.1) as well as the Mixer-to-Client audio levels (Section 5.2.3) (when supported) and the basic requirements for this is further discussed in Section 12.2.1. 12. RTP Implementation Considerations @@ -1303,22 +1217,22 @@ The following discussion provides some guidance on the implementation of the RTP features described in this memo. The focus is on a WebRTC end-point implementation perspective, and while some mention is made of the behaviour of middleboxes, that is not the focus of this memo. 12.1. Configuration and Use of RTP Sessions A WebRTC end-point will be a simultaneous participant in one or more RTP sessions. Each RTP session can convey multiple media sources, and can include media data from multiple end-points. In the - following, we outline some ways in which WebRTC end-points can - configure and use RTP sessions. + following, some ways in which WebRTC end-points can configure and use + RTP sessions is outlined. 12.1.1. Use of Multiple Media Sources Within an RTP Session RTP is a group communication protocol, and every RTP session can potentially contain multiple RTP packet streams. There are several reasons why this might be desirable: Multiple media types: Outside of WebRTC, it is common to use one RTP session for each type of media sources (e.g., one RTP session for audio sources and one for video sources, each sent over different @@ -1397,33 +1311,33 @@ To separate media with different purposes: An end-point might want to send RTP packet streams that have different purposes on different RTP sessions, to make it easy for the peer device to distinguish them. For example, some centralised multiparty conferencing systems display the active speaker in high resolution, but show low resolution "thumbnails" of other participants. Such systems might configure the end-points to send simulcast high- and low-resolution versions of their video using separate RTP sessions, to simplify the operation of the RTP - middlebox. In the WebRTC context this is currently possible to - accomplished by establishing multiple WebRTC MediaStreamTracks - that have the same media source in one (or more) - RTCPeerConnection. Each MediaStreamTrack is then configured to - deliver a particular media quality and thus media bit-rate, and - will produce an independently encoded version with the codec - parameters agreed specifically in the context of that - RTCPeerConnection. The RTP middlebox can distinguish packets - corresponding to the low- and high-resolution streams by - inspecting their SSRC, RTP payload type, or some other information - contained in RTP payload, RTP header extension or RTCP packets, - but it can be easier to distinguish the RTP packet streams if they - arrive on separate RTP sessions on separate transport-layer flows. + middlebox. In the WebRTC context this is currently possible by + establishing multiple WebRTC MediaStreamTracks that have the same + media source in one (or more) RTCPeerConnection. Each + MediaStreamTrack is then configured to deliver a particular media + quality and thus media bit-rate, and will produce an independently + encoded version with the codec parameters agreed specifically in + the context of that RTCPeerConnection. The RTP middlebox can + distinguish packets corresponding to the low- and high-resolution + streams by inspecting their SSRC, RTP payload type, or some other + information contained in RTP payload, RTP header extension or RTCP + packets, but it can be easier to distinguish the RTP packet + streams if they arrive on separate RTP sessions on separate + transport-layer flows. To directly connect with multiple peers: A multi-party conference does not need to use an RTP middlebox. Rather, a multi-unicast mesh can be created, comprising several distinct RTP sessions, with each participant sending RTP traffic over a separate RTP session (that is, using an independent RTCPeerConnection object) to every other participant, as shown in Figure 1. This topology has the benefit of not requiring an RTP middlebox node that is trusted to access and manipulate the media data. The downside is that it increases the used bandwidth at each sender by requiring @@ -1440,37 +1354,37 @@ +---+ | C | +---+ Figure 1: Multi-unicast using several RTP sessions The multi-unicast topology could also be implemented as a single RTP session, spanning multiple peer-to-peer transport layer connections, or as several pairwise RTP sessions, one between each pair of peers. To maintain a coherent mapping between the - relation between RTP sessions and RTCPeerConnection objects we + relation between RTP sessions and RTCPeerConnection objects it is recommend that this is implemented as several individual RTP sessions. The only downside is that end-point A will not learn of the quality of any transmission happening between B and C, since it will not see RTCP reports for the RTP session between B and C, whereas it would it all three participants were part of a single RTP session. Experience with the Mbone tools (experimental RTP- based multicast conferencing tools from the late 1990s) has showed - that RTCP reception quality reports for third parties can usefully - be presented to the users in a way that helps them understand - asymmetric network problems, and the approach of using separate - RTP sessions prevents this. However, an advantage of using - separate RTP sessions is that it enables using different media - bit-rates and RTP session configurations between the different - peers, thus not forcing B to endure the same quality reductions if - there are limitations in the transport from A to C as C will. It - it believed that these advantages outweigh the limitations in + that RTCP reception quality reports for third parties can be + presented to users in a way that helps them understand asymmetric + network problems, and the approach of using separate RTP sessions + prevents this. However, an advantage of using separate RTP + sessions is that it enables using different media bit-rates and + RTP session configurations between the different peers, thus not + forcing B to endure the same quality reductions if there are + limitations in the transport from A to C as C will. It is + believed that these advantages outweigh the limitations in debugging power. To indirectly connect with multiple peers: A common scenario in multi-party conferencing is to create indirect connections to multiple peers, using an RTP mixer, translator, or some other type of RTP middlebox. Figure 2 outlines a simple topology that might be used in a four-person centralised conference. The middlebox acts to optimise the transmission of RTP packet streams from certain perspectives, either by only sending some of the received RTP packet stream to any given receiver, or by providing a @@ -1487,35 +1401,34 @@ Figure 2: RTP mixer with only unicast paths There are various methods of implementation for the middlebox. If implemented as a standard RTP mixer or translator, a single RTP session will extend across the middlebox and encompass all the end-points in one multi-party session. Other types of middlebox might use separate RTP sessions between each end-point and the middlebox. A common aspect is that these RTP middleboxes can use a number of tools to control the media encoding provided by a - WebRTC end-point. This includes functions like requesting - breaking the encoding chain and have the encoder produce a so + WebRTC end-point. This includes functions like requesting the + breaking of the encoding chain and have the encoder produce a so called Intra frame. Another is limiting the bit-rate of a given stream to better suit the mixer view of the multiple down-streams. Others are controlling the most suitable frame-rate, picture resolution, the trade-off between frame-rate and spatial quality. - The middlebox gets the significant responsibility to correctly - perform congestion control, source identification, manage - synchronisation while providing the application with suitable - media optimizations. The middlebox is also has to be a trusted - node when it comes to security, since it manipulates either the - RTP header or the media itself (or both) received from one end- - point, before sending it on towards the end-point(s), thus they - need to be able to decrypt and then encrypt it before sending it - out. + The middlebox has the responsibility to correctly perform + congestion control, source identification, manage synchronisation + while providing the application with suitable media optimisations. + The middlebox also has to be a trusted node when it comes to + security, since it manipulates either the RTP header or the media + itself (or both) received from one end-point, before sending it on + towards the end-point(s), thus they need to be able to decrypt and + then re-encrypt the RTP packet stream before sending it out. RTP Mixers can create a situation where an end-point experiences a situation in-between a session with only two end-points and multiple RTP sessions. Mixers are expected to not forward RTCP reports regarding RTP packet streams across themselves. This is due to the difference in the RTP packet streams provided to the different end-points. The original media source lacks information about a mixer's manipulations prior to sending it the different receivers. This scenario also results in that an end-point's feedback or requests goes to the mixer. When the mixer can't act @@ -1528,21 +1441,21 @@ challenge. In the mixer-based topologies, end-points source authentication is based on, firstly, verifying that media comes from the mixer by cryptographic verification and, secondly, trust in the mixer to correctly identify any source towards the end- point. In RTP sessions where multiple end-points are directly visible to an end-point, all end-points will have knowledge about each others' master keys, and can thus inject packets claimed to come from another end-point in the session. Any node performing relay can perform non-cryptographic mitigation by preventing forwarding of packets that have SSRC fields that came from other - end-points before. For cryptographic verification of the source + end-points before. For cryptographic verification of the source, SRTP would require additional security mechanisms, for example TESLA for SRTP [RFC4383], that are not part of the base WebRTC standards. To forward media between multiple peers: It is sometimes desirable for an end-point that receives an RTP packet stream to be able to forward that RTP packet stream to a third party. The are some obvious security and privacy implications in supporting this, but also potential uses. This is supported in the W3C API by taking the received and decoded media and using it as media source that @@ -1561,21 +1474,21 @@ The end-point that is performing the forwarding is responsible for producing an RTP packet stream suitable for onwards transmission. The outgoing RTP session that is used to send the forwarded media is entirely separate to the RTP session on which the media was received. This will require media transcoding for congestion control purpose to produce a suitable bit-rate for the outgoing RTP session, reducing media quality and forcing the forwarding end-point to spend the resource on the transcoding. The media transcoding does result in a separation of the two different legs removing almost all dependencies, and allowing the forwarding end- - point to optimize its media transcoding operation. The cost is + point to optimise its media transcoding operation. The cost is greatly increased computational complexity on the forwarding node. Receivers of the forwarded stream will see the forwarding device as the sender of the stream, and will not be able to tell from the RTP layer that they are receiving a forwarded stream rather than an entirely new RTP packet stream generated by the forwarding device. 12.1.3. Differentiated Treatment of RTP Packet Streams There are use cases for differentiated treatment of RTP packet @@ -1596,22 +1509,22 @@ retransmission and FEC. The importance of such redundant RTP packet streams is dependent on the media type and codec used, in regards to how robust that codec is to packet loss. However, a default policy might to be to use the same priority for redundant RTP packet stream as for the source RTP packet stream. Secondly, the network can prioritize transport-layer flows and sub- flows, including RTP packet streams. Typically, differential treatment includes two steps, the first being identifying whether an IP packet belongs to a class that has to be treated differently, the - second the actual mechanism to prioritize packets. This is done - according to three methods: + second consisting of the actual mechanism to prioritize packets. + This is done according to three methods: DiffServ: The end-point marks a packet with a DiffServ code point to indicate to the network that the packet belongs to a particular class. Flow based: Packets that need to be given a particular treatment are identified using a combination of IP and port address. Deep Packet Inspection: A network classifier (DPI) inspects the packet and tries to determine if the packet represents a @@ -1639,39 +1552,40 @@ and are further discussed in "DSCP and other packet markings for RTCWeb QoS" [I-D.ietf-tsvwg-rtcweb-qos]. For packet based marking schemes it might be possible to mark individual RTP packets differently based on the relative priority of the RTP payload. For example video codecs that have I, P, and B pictures could prioritise any payloads carrying only B frames less, as these are less damaging to loose. However, depending on the QoS mechanism and what markings that are applied, this can result in not only different packet drop probabilities but also packet reordering, - see [I-D.ietf-tsvwg-rtcweb-qos] for further discussion. As default + see [I-D.ietf-tsvwg-rtcweb-qos] for further discussion. As a default policy all RTP packets related to a RTP packet stream ought to be provided with the same prioritization; per-packet prioritization is outside the scope of this memo, but might be specified elsewhere in future. It is also important to consider how RTCP packets associated with a particular RTP packet stream need to be marked. RTCP compound packets with Sender Reports (SR), ought to be marked with the same priority as the RTP packet stream itself, so the RTCP-based round- trip time (RTT) measurements are done using the same transport-layer flow priority as the RTP packet stream experiences. RTCP compound packets containing RR packet ought to be sent with the priority used by the majority of the RTP packet streams reported on. RTCP packets containing time-critical feedback packets can use higher priority to improve the timeliness and likelihood of delivery of such feedback. 12.2. Media Source, RTP Packet Streams, and Participant Identification -12.2.1. Media Source + +12.2.1. Media Source Identification Each RTP packet stream is identified by a unique synchronisation source (SSRC) identifier. The SSRC identifier is carried in each of the RTP packets comprising a RTP packet stream, and is also used to identify that stream in the corresponding RTCP reports. The SSRC is chosen as discussed in Section 4.8. The first stage in demultiplexing RTP and RTCP packets received on a single transport layer flow at a WebRTC end-point is to separate the RTP packet streams based on their SSRC value; once that is done, additional demultiplexing steps can determine how and where to render the media. @@ -1690,70 +1604,72 @@ indicates which participants are active in the session. Changes in the CSRC list included in packets needs to be exposed to the WebRTC application using some API, if the application is to be able to track changes in session participation. It is desirable to map CSRC values back into WebRTC MediaStream identities as they cross this API, to avoid exposing the SSRC/CSRC name space to JavaScript applications. If the mixer-to-client audio level extension [RFC6465] is being used in the session (see Section 5.2.3), the information in the CSRC list is augmented by audio level information for each contributing source. - This information can usefully be exposed in the user interface. + It is desirable to expose this information to the WebRTC application + using some API, after mapping the CSRC values to WebRTC MediaStream + identities, so it can be exposed in the user interface. 12.2.2. SSRC Collision Detection - The RTP standard [RFC3550] requires any RTP implementation to have - support for detecting and handling SSRC collisions, i.e., resolve the - conflict when two different end-points use the same SSRC value. This - requirement also applies to WebRTC end-points. There are several - scenarios where SSRC collisions can occur: + The RTP standard requires RTP implementations to have support for + detecting and handling SSRC collisions, i.e., resolve the conflict + when two different end-points use the same SSRC value (see section + 8.2 of [RFC3550]). This requirement also applies to WebRTC end- + points. There are several scenarios where SSRC collisions can occur: o In a point-to-point session where each SSRC is associated with either of the two end-points and where the main media carrying SSRC identifier will be announced in the signalling channel, a collision is less likely to occur due to the information about - used SSRCs provided by Source-Specific SDP Attributes [RFC5576]. - - Still, collisions can occur if both end-points start uses an new - SSRC identifier prior to having signalled it to the peer and - received acknowledgement on the signalling message. The Source- - Specific SDP Attributes [RFC5576] contains no mechanism to resolve - SSRC collisions or reject a end-points usage of an SSRC. + used SSRCs. If SDP is used, this information is provided by + Source-Specific SDP Attributes [RFC5576]. Still, collisions can + occur if both end-points start using a new SSRC identifier prior + to having signalled it to the peer and received acknowledgement on + the signalling message. The Source-Specific SDP Attributes + [RFC5576] contains a mechanism to signal how the end-point + resolved the SSRC collision. o SSRC values that have not been signalled could also appear in an RTP session. This is more likely than it appears, since some RTP functions use extra SSRCs to provide their functionality. For example, retransmission data might be transmitted using a separate RTP packet stream that requires its own SSRC, separate to the SSRC of the source RTP packet stream [RFC4588]. In those cases, an end-point can create a new SSRC that strictly doesn't need to be announced over the signalling channel to function correctly on both RTP and RTCPeerConnection level. o Multiple end-points in a multiparty conference can create new sources and signal those towards the RTP middlebox. In cases where the SSRC/CSRC are propagated between the different end- points from the RTP middlebox collisions can occur. o An RTP middlebox could connect an end-point's RTCPeerConnection to another RTCPeerConnection from the same end-point, thus forming a - loop where the end-point will receive its own traffic. While is + loop where the end-point will receive its own traffic. While it is clearly considered a bug, it is important that the end-point is able to recognise and handle the case when it occurs. This case becomes even more problematic when media mixers, and so on, are involved, where the stream received is a different stream but still contains this client's input. These SSRC/CSRC collisions can only be handled on RTP level as long as the same RTP session is extended across multiple RTCPeerConnections by a RTP middlebox. To resolve the more generic - case where multiple RTCPeerConnections are interconnected, then + case where multiple RTCPeerConnections are interconnected, identification of the media source(s) part of a MediaStreamTrack being propagated across multiple interconnected RTCPeerConnection needs to be preserved across these interconnections. 12.2.3. Media Synchronisation Context When an end-point sends media from more than one media source, it needs to consider if (and which of) these media sources are to be synchronized. In RTP/RTCP, synchronisation is provided by having a set of RTP packet streams be indicated as coming from the same @@ -1773,22 +1689,22 @@ 13. Security Considerations The overall security architecture for WebRTC is described in [I-D.ietf-rtcweb-security-arch], and security considerations for the WebRTC framework are described in [I-D.ietf-rtcweb-security]. These considerations also apply to this memo. The security considerations of the RTP specification, the RTP/SAVPF profile, and the various RTP/RTCP extensions and RTP payload formats that form the complete protocol suite described in this memo apply. - We do not believe there are any new security considerations resulting - from the combination of these various protocol extensions. + It is not believed there are any new security considerations + resulting from the combination of these various protocol extensions. The Extended Secure RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback [RFC5124] (RTP/SAVPF) provides handling of fundamental issues by offering confidentiality, integrity and partial source authentication. A mandatory to implement media security solution is created by combing this secured RTP profile and DTLS-SRTP keying [RFC5764] as defined by Section 5.5 of [I-D.ietf-rtcweb-security-arch]. RTCP packets convey a Canonical Name (CNAME) identifier that is used @@ -1816,23 +1731,24 @@ 14. IANA Considerations This memo makes no request of IANA. Note to RFC Editor: this section is to be removed on publication as an RFC. 15. Acknowledgements The authors would like to thank Bernard Aboba, Harald Alvestrand, - Cary Bran, Charles Eckel, Christian Groves, Cullen Jennings, Dan - Romascanu, Martin Thomson, and the other members of the IETF RTCWEB - working group for their valuable feedback. + Cary Bran, Ben Campbell, Charles Eckel, Alex Eleftheriadis, Christian + Groves, Cullen Jennings, Olle Johansson, Suhas Nandakumar, Dan + Romascanu, Jim Spring, Martin Thomson, and the other members of the + IETF RTCWEB working group for their valuable feedback. 16. References 16.1. Normative References [I-D.ietf-avtcore-multi-media-rtp-session] Westerlund, M., Perkins, C., and J. Lennox, "Sending Multiple Types of Media in a Single RTP Session", draft- ietf-avtcore-multi-media-rtp-session-05 (work in progress), February 2014. @@ -1951,34 +1867,20 @@ [RFC7022] Begen, A., Perkins, C., Wing, D., and E. Rescorla, "Guidelines for Choosing RTP Control Protocol (RTCP) Canonical Names (CNAMEs)", RFC 7022, September 2013. [RFC7160] Petit-Huguenin, M. and G. Zorn, "Support for Multiple Clock Rates in an RTP Session", RFC 7160, April 2014. [RFC7164] Gross, K. and R. Brandenburg, "RTP and Leap Seconds", RFC 7164, March 2014. - [W3C.WD-mediacapture-streams-20130903] - Burnett, D., Bergkvist, A., Jennings, C., and A. - Narayanan, "Media Capture and Streams", World Wide Web - Consortium WD WD-mediacapture-streams-20130903, September - 2013, . - - [W3C.WD-webrtc-20130910] - Bergkvist, A., Burnett, D., Jennings, C., and A. - Narayanan, "WebRTC 1.0: Real-time Communication Between - Browsers", World Wide Web Consortium WD WD- - webrtc-20130910, September 2013, - . - 16.2. Informative References [I-D.ietf-avtcore-multiplex-guidelines] Westerlund, M., Perkins, C., and H. Alvestrand, "Guidelines for using the Multiplexing Features of RTP to Support Multiple Media Streams", draft-ietf-avtcore- multiplex-guidelines-02 (work in progress), January 2014. [I-D.ietf-avtcore-rtp-topologies-update] Westerlund, M. and S. Wenger, "RTP Topologies", draft- @@ -2031,59 +1933,58 @@ [I-D.ietf-tsvwg-rtcweb-qos] Dhesikan, S., Druta, D., Jones, P., and J. Polk, "DSCP and other packet markings for RTCWeb QoS", draft-ietf-tsvwg- rtcweb-qos-00 (work in progress), April 2014. [RFC3611] Friedman, T., Caceres, R., and A. Clark, "RTP Control Protocol Extended Reports (RTCP XR)", RFC 3611, November 2003. - [RFC4341] Floyd, S. and E. Kohler, "Profile for Datagram Congestion - Control Protocol (DCCP) Congestion Control ID 2: TCP-like - Congestion Control", RFC 4341, March 2006. - - [RFC4342] Floyd, S., Kohler, E., and J. Padhye, "Profile for - Datagram Congestion Control Protocol (DCCP) Congestion - Control ID 3: TCP-Friendly Rate Control (TFRC)", RFC 4342, - March 2006. - [RFC4383] Baugher, M. and E. Carrara, "The Use of Timed Efficient Stream Loss-Tolerant Authentication (TESLA) in the Secure Real-time Transport Protocol (SRTP)", RFC 4383, February 2006. - [RFC4828] Floyd, S. and E. Kohler, "TCP Friendly Rate Control - (TFRC): The Small-Packet (SP) Variant", RFC 4828, April - 2007. - - [RFC5348] Floyd, S., Handley, M., Padhye, J., and J. Widmer, "TCP - Friendly Rate Control (TFRC): Protocol Specification", RFC - 5348, September 2008. + [RFC5245] Rosenberg, J., "Interactive Connectivity Establishment + (ICE): A Protocol for Network Address Translator (NAT) + Traversal for Offer/Answer Protocols", RFC 5245, April + 2010. [RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific Media Attributes in the Session Description Protocol (SDP)", RFC 5576, June 2009. - [RFC5681] Allman, M., Paxson, V., and E. Blanton, "TCP Congestion - Control", RFC 5681, September 2009. - [RFC5968] Ott, J. and C. Perkins, "Guidelines for Extending the RTP Control Protocol (RTCP)", RFC 5968, September 2010. [RFC6263] Marjou, X. and A. Sollaud, "Application Mechanism for Keeping Alive the NAT Mappings Associated with RTP / RTP Control Protocol (RTCP) Flows", RFC 6263, June 2011. [RFC6792] Wu, Q., Hunt, G., and P. Arden, "Guidelines for Use of the RTP Monitoring Framework", RFC 6792, November 2012. + [W3C.WD-mediacapture-streams-20130903] + Burnett, D., Bergkvist, A., Jennings, C., and A. + Narayanan, "Media Capture and Streams", World Wide Web + Consortium WD WD-mediacapture-streams-20130903, September + 2013, . + + [W3C.WD-webrtc-20130910] + Bergkvist, A., Burnett, D., Jennings, C., and A. + Narayanan, "WebRTC 1.0: Real-time Communication Between + Browsers", World Wide Web Consortium WD WD- + webrtc-20130910, September 2013, + . + Authors' Addresses Colin Perkins University of Glasgow School of Computing Science Glasgow G12 8QQ United Kingdom Email: csp@csperkins.org URI: http://csperkins.org/